Method and apparatus for processing an audio signal based on an estimated loudness

ABSTRACT

An apparatus comprising at least one processor and at least one memory including computer program code. The at least one memory and the computer program code is configured to, with the at least one processor, cause the apparatus at least to determine a loudness estimate of a first audio signal, generate a parameter dependent on the loudness estimate; and control the first audio signal dependent on the parameter.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. application Ser. No.13/697,425, filed Feb. 4, 2013, which is a national phase entry ofInternational Application No. PCT/IB2010/052113 filed May 12, 2010, theentire contents of which are incorporated herein by reference.

The present invention relates to apparatus for processing of audiosignals. The invention further relates to, but is not limited to,apparatus for processing audio and speech signals in audio playbackdevices.

Audio processing and in particular audio processing in mobile deviceshave been a growing area in recent years.

The use of portable audio playback devices is therefore becoming common.It would be desirable that the loudness experienced by the user isconstant or nearly constant independent of the source of the audio. Forexample, it can be common experience that speech audio during a phonecall may have a significantly lower perceived loudness than a precedingmusic audio signal, which causes the user to increase the volume levelin order to increase the loudness of the voice. However when the phonecall ends and the user returns to listening to music with a higherloudness, this event can startle the user and require the user to reducethe volume.

In general this situation is not purely volume related but is perceptionbased. A source signal can be perceived loud at a specific volume levelwhereas an alternative source signal can be perceived to cause lessloudness at a same volume level. For example as well as the above voiceand music switch different types of music file from their library cancause similar problems. In other words the user may change the volumelevel as the device switches between a classical music audio source,rock audio source, or pop audio source.

This invention thus proceeds from the consideration that by using aloudness model to estimate a perceived loudness of the signal gain orsimilar digital signal processing parameters may be adoptively adjusted.

Embodiments of the present invention aim to address the above problem.

There is provided according to a first aspect of the invention a methodcomprising: a method comprising: determining a loudness estimate of afirst audio signal; generating a parameter dependent on the loudnessestimate; and controlling the first audio signal dependent on theparameter.

Determining a loudness estimate may comprise applying at least oneloudness model to the first audio signal;

The at least one loudness model may comprise at least one of; a digitalfilter; and a parametric filter.

The method may further comprise determining whether the first audiosignal is speech, wherein the determining the loudness estimate of thefirst audio signal is dependent on the determination of whether thefirst audio signal is speech.

The at least one loudness model may comprise a speech loudness model tobe applied to the first audio signal when the first audio signal isspeech.

Determining the loudness estimate may further comprise applying anenvironmental model.

The environmental model may comprise at least one of: a loudspeakermodel comprising at least one filter simulating the audio filtering ofan integrated hands free loudspeaker; an earpiece model comprising atleast one filter simulating the audio filtering of an earpiecetransducer and housing; a headset model comprising at least one filtersimulating the audio filtering of a predetermined headset; and amechanical model comprising at least one filter simulating themechanical acoustical response of the apparatus.

The method may further comprise smoothing the loudness estimate.

The parameter may comprise at least one of: a gain control parameterapplied to an amplifier configured to control the first audio signal; adynamic range parameter applied to a dynamic range processor configuredto control the first audio signal; and a dynamic level parameter appliedto a dynamic level controller configured to control the first audiosignal.

Generating a parameter depends on the loudness estimate may comprise:comparing the loudness estimate with a loudness input value; andgenerating the parameter dependent on the difference between theloudness estimate of the first audio signal and the loudness inputvalue.

According to a second aspect of the application there is provided anapparatus comprising at least one processor and at least one memoryincluding computer program code the at least one memory and the computerprogram code configured to, with the at least one processor, cause theapparatus at least to perform; determining a loudness estimate of afirst audio signal; generating a parameter dependent on the loudnessestimate; and controlling the first audio signal dependent on theparameter.

The apparatus caused to perform determining a loudness estimate maycause the apparatus to apply at least one loudness model to the firstaudio signal;

The at least one loudness model may comprise at least one of: a digitalfiler; and a parametric filter.

The apparatus may be further caused to perform determining whether thefirst audio signal is speech, wherein the determining the loudnessestimate of the first audio signal is dependent on the determination ofwhether the first audio signal is speech.

The at least one loudness model may comprise a speech loudness model tobe applied to the first audio signal when the first audio signal isspeech.

The apparatus caused to perform determining the loudness estimate may befurther caused to perform applying an environmental model.

The environmental model may comprise at least one of: a loudspeakermodel comprising at least one filter simulating the audio filtering ofan integrated hands free loudspeaker; an earpiece model comprising atleast one filter simulating the audio filtering of an earpiecetransducer and housing; a headset model comprising at least one filtersimulating the audio filtering of a predetermined headset; and amechanical model comprising at least one filer simulating the mechanicalacoustical response of the apparatus.

The apparatus may be further caused to perform smoothing the loudnessestimate.

The parameter may comprise at least one of: a gain control parameterapplied to an amplifier configured to control the first audio signal; adynamic range parameter applied to a dynamic range processor configuredto control the first audio signal; and a dynamic level parameter appliedto a dynamic level controller configured to control the first audiosignal.

The apparatus caused to perform generating a parameter dependent on theloudness estimate may be further caused to perform: comparing theloudness estimate with a loudness input value; and generating theparameter dependent on the difference between the loudness estimate ofthe first audio signal and the loudness input value.

According to a third aspect of the invention there is provided anapparatus comprising: a loudness estimator configured to determine aloudness estimate of a first audio signal; a parameter generatorconfigured to generate a parameter dependent on the loudness estimate;and a signal conditioner configured to control the first audio signaldependent on the parameter.

The loudness estimator may comprise a signal processor configured toapply at least one loudness model to the first audio signal;

The at least one loudness model may comprise at least one of: a digitalfilter; and a parametric filter.

The apparatus may further comprise a speech determiner configured todetermine whether the first audio signal is speech, wherein the loudnessestimator may be configured to estimate the loudness of the first audiosignal dependent on the output of the speech determiner.

The loudness estimator may be configured to apply a speech loudnessmodel to the first audio signal when speech determiner determiners thefirst audio signal is speech.

The loudness estimator may be configured to apply an environmental modelto determine the loudness estimate.

The environmental model may comprise at least one of: a loudspeakermodel comprising at least one filter simulating the audio filtering ofan integrated hands free loudspeaker; an earpiece model comprising atleast one filter simulating the audio filtering of an earpiecetransducer and housing; a headset model comprising at least one filtersimulating the audio filtering of a predetermined headset; and amechanical model comprising at least one filter simulating themechanical acoustical response of the apparatus.

The apparatus may further comprise a low pass filter configured tosmooth the loudness estimate.

The parameter may comprise at least one of: a gain control parameterapplied to an amplifier configured to control the first audio signal; adynamic range parameter applied to a dynamic range processor configuredto control the first audio signal; and a dynamic parameter applied to adynamic level controller configured to control the first audio signal.

The parameter generator may comprise: a loudness comparator configuredto compare the loudness estimate with a loudness input value; and aparameter determiner configured to generate the parameter dependent onthe difference between the loudness estimate of the first audio signaland the loudness input value.

The signal conditioner may comprise an amplifier; a dynamic rangeprocessor; and a dynamic level controller.

According to a fourth aspect of the invention there is provided anapparatus comprising: loudness estimator means configured to determine aloudness estimate of a first audio signal; parameter generator meansconfigured to generate a parameter dependent on the loudness estimate;and signal conditioner means configured to control the first audiosignal dependent on the parameter.

According to a fifth aspect of the invention there is provided acomputer-readable medium encoded with instructions that, when executedby a computer perform: determining a loudness estimate of a first audiosignal; generating a parameter dependent on the loudness estimate; andcontrolling the first audio signal dependent on the parameter.

An electronic device may comprise apparatus as described above.

A chipset may comprise apparatus as described above.

BRIEF DESCRIPTION OF DRAWINGS

For better understanding of the present invention, reference will now bemade by way of example to the accompanying drawings in which:

FIG. 1 shows schematically an apparatus employing embodiments of theapplication;

FIG. 2 shows schematically apparatus suitable for implementing someembodiments of the application;

FIG. 3 shows schematically the loudness estimator as shown in FIG. 2 infurther detail;

FIG. 4 shows schematically some further apparatus suitable forimplementing some embodiments of the application;

FIG. 5 shows a parameter updater as shown in FIGS. 2 and 4 in furtherdetail;

FIG. 6 shows a speech detector as shown in FIGS. 2 and 4 in furtherdetail;

FIG. 7 shows a flow diagram showing the overview operation of someembodiments of the application;

FIG. 8 shows a flow diagram showing the operation of the speech detectoraccording to some embodiments of the application;

FIG. 9 shows a flow diagram showing the operation of the loudnessestimator to some embodiments of the application;

FIG. 10 shows a flow diagram showing the operation of the parameterupdater according to embodiments of the application; and

FIG. 11 shows a flow diagram showing the operation of a feedback loopshown in the further apparatus as shown in FIG. 4.

The following describes apparatus and methods for the provision ofenhancing audio signal processing and playback. In this regard referenceis first made to FIG. 1 which shows a schematic block diagram of anexemplary electronic device 10 or apparatus, which may incorporateplayback apparatus according to some embodiments of the application.

The apparatus 10 may for example be a mobile terminal or user equipmentfor a wireless communication system. In other embodiments the electronicdevice may be a Television (TV) receiver, portable digital versatiledisc (DVD) player, an audio player such as an mp3 player, or mediaplayer such as a mp4 player.

The apparatus 10 comprises a processor 21 which may be linked via adigital-to-analogue converter 32 to a playback speaker configured toprovide a suitable audio playback. The playback speaker in someembodiments may be any suitable loudspeaker. In some other embodimentsthe playback speaker may be a headphone or ear worn speaker (EWS) set.In some embodiments the apparatus 10 may comprise a headphone connectorfor receiving a headphone or headset 33. The processor 21 is in someembodiments further linked to a transceiver (TX/RX) 13, to a userinterface (UI) 15 and to a memory 22.

The processor 21 may be configured to execute various program codes. Theimplemented program codes comprise a loudness balancer for processing anaudio signal and thus balance the loudness to the desired level. Theimplemented program codes 23 may be stored for example in the memory 22for retrieval by the processor 21 whenever needed. The memory 22 couldfurther provide a section 24 for storing data, for example data that hasbeen processed in accordance with the embodiments.

The loudness balancing code may in embodiments be implemented inhardware or firmware. In the following example a schematic hardwareimplementation is described however it would be understood that variousprogram codes, stored for example in memory 22, can also implement theloudness balancing operation.

The user interface 15 in some embodiments enables a user to inputcommands to the electronic device 10, for example via a keypad, and/orto obtain information from the electronic device 10, for example via adisplay. The transceiver 13 enables a communication with otherelectronic devises, for example via a wireless communication network.

It is to be understood again that the structure of the electronic device10 could be supplemented and varied in many ways.

The apparatus 10 can in some embodiments further comprise at least onemicrophone 11 for monitoring audio or speech that is to be processedaccording to embodiments of the application. A corresponding applicationto capture audio signals using the microphone may be activated to thisend by the user via the user interface 15. The apparatus 10 in suchembodiments may further comprise an analogue-to-digital converter 14configured to convert the input analogue audio signal into a digitalaudio signal and provide the digital audio signal to the processor 21.

The apparatus 10 may in some embodiments receive a bit stream with acorrespondingly encoded audio data from another electronic device viathe transceiver 13. In these embodiments, the processor 21 may executethe processing program code stored in the memory 22. The processor 21 inthese embodiments may process the received audio signal data, and outputthe processed audio.

In some embodiments the headphone connector 33 may be configured tocommunicate to a headphone set or earplugs wirelessly, for example by aBluetooth profile, or using a conventional wired connection.

The received stereo audio data may in some embodiments also be stored,instead of being processed immediately, in the data section 24 of thememory 22, for instance for enabling a later processing and presentationor forwarding to still another electronic device.

It would be appreciated that the schematic structures described in FIGS.2 to 6 and the method steps shown in FIG. 7 to 11 represent only a partof the operation of a complete audio processing chain comprising someembodiments as exemplary the shown implemented in the apparatus shown inFIG. 1.

With respect to FIG. 2, a schematic view of the loudness balancer 199.The following examples describe a single channel loudness balancer,however, it would be appreciated that in other embodiments of theapplication, multiple channel balancers may be applied whereby eachchannel has its own balancer or a mixer used to generate a mixed channelfor analysis to produce an overall balancing effect to be applied toeach channel could be implemented.

In some embodiments of the application the loudness balancer 199comprises a segmenter 100 configured to segment the received audiosignals on one pathway, which may be called the analysis pathway, and aprocessing component on a second pathway, also known as the processingpath or signal pathway. The input to the loudness balancer 199 is thusin these embodiments is passed to the segmenter 100 on the analysispathway and also to a gain amplifier 109 on the signal pathway. It wouldbe appreciated that in some embodiments the signal pathway may be anysuitable processing of the signal required to be carried out. Forexample in some embodiments the signal pathway may comprise at least oneof an analogue and digital processor suitable for processing the inputsignal and also configured to receive at least one control input fromthe analysis pathway.

The segmenter 100 receives the audio signal in a digital form andconverts the audio signal into a sequence of windowed frames each of aspecific length, period, or number of samples long. For example in someembodiments the segment/window generated by the segmenter 100 may have a40 millisecond (ms) window length. Furthermore in some embodiments thesegmenter 100 generates a segment (or frame or window) every 20milliseconds. In other words a new segment is started after 20milliseconds and each segment overlaps by 20 milliseconds with theprevious segment and by 20 milliseconds with the next segment. It wouldbe understood that any suitable window size and repetition may be used,for example in some further embodiments the window size may be from 20to 50 milliseconds long and have a similar range of overlaps from 20 to50 ms.

The segmentation of the input signal into frames is shown in FIG. 7 bystep 601. Each segmented frame may then be passed to the loudnessestimator 103 and the speech detector 101.

In some embodiments the loudness balancer 199 further comprises a speechdetector 101, which is configured to analyse the frames to determine ifthe frame if speech frame or a non-speech frame. In such embodiments thespeech detector is configured to analyse the audio signal to determinewhen frames are speech and non-speech or silent. In some embodiments forexample where the audio signal is a downlink signal (an example of whichis a cellular or mobile call downlink to the apparatus) where speech maybe present it is known that typical speech can be divided into speechframes comprising energy partials or formants where harmonic relationscan be found and silent portions. Where a loudness estimate is to begenerated according to some embodiments the loudness estimate isdetermined in relation to the speech frames and not the whole audiosignal as the whole audio signal may provide a low estimate. In someembodiments the speech detector 101 is inactive or not present.

In some embodiments thus the speech detector 101 therefore receives thesegmented frames from the segmenter 100 and is configured to determinewhether each frame is a speech audio signal or a non-speech audio signaland pass the result of this determination to the loudness estimator 103.The operation of determining whether or not the frame is speech ornon-speech is shown in FIG. 7 by step 603. It would be understood thatin some embodiments that is operation would not be performed.

With respect to FIG. 3, a speech detector 101 according to someembodiments of the application is shown in further detail. Furthermorewith respect to FIG. 8 the operation of the speech detector 101according to embodiments of the application is shown in further detail.

The speech detector 101 in some embodiments comprises a time tofrequency domain converter 501 which receives the frames from thesegment window generator 100 and converts the time domain samples intofrequency domain representations. The time to frequency domain converter501 in some embodiments comprises any of a Fast Fourier Transformer, amodified Discrete Cosine Transformer (MDCT), a Wavelet Transformer,Discrete Fourier Transform, Short-Term Fast Fourier Transform (FFT),Goertzel's algorithm or any suitable time to frequency domain converter.In some embodiments the time to frequency domain converter 501 isconfigured to output only real frequency domain components whereas inother embodiments, both real and imaginary components may be output inorder to preserve both amplitude and phase information. In someembodiments a zero padding operation can further be applied to improvethe accuracy of the transform. The frequency domain components may thenbe output to the sub-bank peak detector 503 and a peak energy determiner505.

The generation of frequency domain components is shown in FIG. 8 by step701.

In some embodiments the speech detector further comprises a sub-bandpeak detector 503. The sub-band peak detector 503 may be configured insome embodiments to compare the received frequency domain components(for example the N sub-bands output from the time to frequency domainconverter 501) and output an indication or signals indicating the peakactivity in the frequency domain components or sub-bands. The sub-bandpeak detector 503 in some embodiments determines a peak sub-band bydetermining the derivative of the sub-band coefficient value andchoosing a sub-band as a peak sub-band where the derivative signswitches before and after the sub-band. For example the sub-bands may beprocessed to calculate the difference between neighbouring sub-bands anda sub-band selected as being a potential peak value if the sign of thedifference between the current sub-band and the previous sub-band isdifferent from the sign of the difference between the next sub-band andthe current sub-band. In some embodiments only a predetermined number ofsub-bands are selected to be passed to the peak comparator. These arethe sub-bands with the N (where N is a predetermined number of sub-bandsto be stored) highest energy sub-bands for the current segment or framethat has been determined to be peak sub-bands.

The peak indicator values may be passed in some embodiments to thesegment peak comparator and to the peak energy determiner 505.

Furthermore in some embodiments the sub-band peak detector 503calculates the energy of each of the peak sub-bands.

The determination of the peak sub-bands is shown in FIG. 8 by step 703.

In some embodiments the speech detector 101 further comprises a peakenergy determiner 505.

The peak energy determiner 505 in some embodiments receives the peaksub-band indicator values representing frequencies where there isactivity from the sub-band peak detector 503 and also the frequencydomain components of the segment. The peak energy determiner 505 thencalculates the ratio of energy at the detected peak or fundamentalfrequency and its harmonics compared to the total energy of the segment.

The determination of the peak sub-band energy is shown in FIG. 8 by step705.

In some embodiments the speech detector 101 further comprises a peakenergy determiner 505.

The segment peak comparator/speech determiner 507 receives in someembodiments both the indications of the peak sub-band activity and alsothe peak energy values from the sub-band peak detector 503 and ratiopeak energy determiner 505 respectively and compares these valuesagainst previous frame segment peak values.

Furthermore in some embodiments the segment peak comparator 507 comparesthe peak values against the energy value for the frame to determinewhether or not the peak value is a dominant sub-band value. The dominantsub-band peaks are then compared against the previous segment frames.The action of comparing the peaks against the total energy of the framein order to determine dominant sub-bands is shown in FIG. 8 by step 707.

The segment peak comparator for example compares the current indicatedpeak values which are determined to be dominant against the previous Msegment peak values to determine whether or not any of the current peakvalues is substantially similar to a previous peak sub-band value and/orhow many previous peak sub-bands is it similar to.

The operation of comparing peak sub-band values against a range ofprevious peak sub-bands is shown in FIG. 8 by step 709.

Where a current segment peak value is similar to a range of previoussegment peak sub-band values then this current segment peak value isselected as a potential fundamental frequency candidate and output tothe fundamental frequency selector. In other words if each sub-band hasan index value i then where at least one of the current segment peakvalue sub-band index values {i_(t)} is within a range of M_(range) ofindex values of previous segment peak values {i_(t-)} then the sub-bandindex-either current or previous peak values is noted. In someembodiments the search range is split equally so that for each of thecurrent sub-band peak values a range of previous peak values of[i_(t)−M_(range)/2, i_(t)+M_(range)/2] is searched. This is becausespeech may have transient forms where the fundamental frequency slidesover the same syllable. The operation of selecting substantially similarpeak sub-bands over a range of segments may be implemented using alinked list memory where each element in the linked list comprises asub-band index value and integer value representing the number ofconsecutive segments the sub-band has been “active” for. In suchembodiments for each segment the segment peak comparator 507 comparesthe current “active” or peak sub-band index value against the list. If acurrent peak sub-band index is outside the search range then the list isincremented by an additional element with the current peak sub-bandindex and a consecutive segment value of 1. If the current peak sub-bandindex is within the search range then the list entries within the searchrange are amended so that there sub-band index value is the currentindex value and the consecutive segment value incremented by 1. In suchembodiments where 2 or more search range generates more than one match,then the entry with the largest consecutive index is chosen. Finally thelist is then pruned to remove all entries which have not been updated oradded in the current segment.

In some embodiments other suitable search and memory operations may beused to monitor consecutive segment sub-bands. Thus the list contains aseries of entries of frequency indicators and the number of consecutivesegments within which the frequency has been a “peak” value.

The segment peak comparator/speech determiner 507 may in someembodiments output a binary decision of whether or not there are anyspeech components within the segment where the number of activeconsecutive sub-bands is within the predefined range of values. Forexample in some embodiments the range of values is defined as beingequal to or greater than a predefined value. For example in someembodiments the segment peak comparator speech determiner 507 maydetermine that speech has been detected where there is at least one peaksub-band over two consecutive segments. In such embodiments the speechdeterminer 507 can output a binary decision S_(speech) of 1.

The operation of comparing the current segment frame against previoussegment frames to determine consecutive values is shown in FIG. 8 bystep 709.

In some other embodiments the predefined range is defined by apredefined lower and upper level, in other words that the number ofconsecutive “active” segments for a sub-band is between a minimum numberand a maximum number.

The operation of determining whether or not the consecutive value isgreater than the predefined value is shown in FIG. 8 by step 711.

Furthermore step 713 shows that when the segment is determined as beingspeech, a speech indicator is output. For example S_(speech) is set to avalue of 1.

Within step 715 where the consecutive value is not greater than thepredefined value according to the comparison step of 711 then the then anon-speech indicator is output. For example S_(speech) is set to a valueof 0.

It would be appreciated that speech/non-speech detection in somaembodiments can be carried out in any other suitable manner. In someembodiments for example all fundamental frequencies within a definedrange (defined by the typical speech signal fundamental frequencies) aremonitored, in other words the dominant sub-bands over a specificfrequency range only are used.

It would also be understood that it would be possible for in someembodiments the speech detector 101 to estimate the fundamentalfrequency and check whether or not it lies within an expected range ofspeech frequencies. As has been indicated previously the output of thespeech detector 101 may in some embodiments be passed to the loudnessestimator 103 to control the loudness estimator in such a way that theloudness estimator is configured only to generate a loudness estimatefor a frame when the frame is determined to be speech but not when theframe is determined to be non-speech or silence in order that theloudness estimate does not inaccurately bias any averaging of theloudness estimate with “silence” loudness estimates.

In some embodiments the loudness balancer 199 further comprises aloudness estimator 103.

The segmented frame data is received by the loudness estimator 103. Insome embodiments, as described above such as the call downlinkembodiment, the loudness estimator 103 may also receive aspeech/non/speech indicator indicating whether or not the currentsegment/frame is speech or non-speech. As described above in suchembodiments where the loudness estimator 103 receives aspeech/non-speech indicator the loudness estimator may be configured toonly produce a loudness estimate for a frame where the speech/non-speechindicator has a speech indication value.

In some embodiments the loudness estimator 103 receives a furtherindicator which has a value representing the environment within whichthe apparatus is operating. For example in some embodiments the loudnessestimator 103 receives an input indicating whether or not the device isbeing used with headphones, for example Bluetooth headphones or wiredconnection headphones, in a hands-free mode and thus using theintegrated hands-free loudspeaker, or being used close to the user's earusing the earpiece loudspeaker within the apparatus. In some embodimentsthis indicator/indicator value is determined by the loudness estimator103, for example the loudness estimator may perform a headphone/headsetdetermination operation determining if a headphone/headset is connectedto the apparatus and in some embodiments what type or model theheadphone/headset is. Furthermore the loudness estimator 103 in someembodiments may determine whether the apparatus is being used in handsfree or earpiece mode from the apparatus proximity detector.

The loudness estimator 103 having received the segment frame isconfigured to estimate a loudness value of the segment/frame within atleast one environment. The loudness estimator 103 as described above insome embodiments receives a input indicating which specific mode orenvironment the loudness estimate is to be determined for.

In some embodiments the loudness estimator 103 produces a loudnessestimate value for more than one environmental situation or mode. Insome of these embodiments each of the loudness estimates are output,whereas in some other embodiments the loudness estimator furthercomprises a switch 202 which enables the loudness estimator 103 toenable a specific pathway and output one loudness estimate associatedwith the environment/mode of operation.

The operation of generating at least one loudness estimate for thesegment frame is shown with regards to FIG. 7 by step 605.

With respect to FIG. 4 which shows a loudness estimator 103 in furtherdetail and FIG. 9 which shows the operation of such a loudnessestimator, the structure and operation of a loudness estimator 103according to some embodiments is further described and explained. In thefollowing discussion of some of the embodiments of the application theloudness estimator 103 comprises a loudness module processor 201 whichoutputs a primary estimate of the loudness which may be passed to aswitch to select one of three displayed environmental/mode estimatemodifiers. It would be appreciated that in some embodiments the loudnessestimator 103 in some embodiments does not comprise a switch and thus asdescribed above can determine and output more than one loudness estimatewhere each loudness estimate is dependent on the environmentalcondition/mode of operation of the apparatus.

The loudness estimator 103 thus in some embodiments comprises a loudnessmodule processor 201. The loudness model processor 201 according to someembodiments simulates the properties of the human auditory system. Insome embodiments the loudness model processor 201 comprises a filter orset of filters which correspond to a selected Equal-Loudness-Curve (ELC)and a mapping from signal energy to loudness. In some embodiments theloudness model processor 201 can apply a known loudness model, forexample the, or any suitable loudness model.

It would be understood that in some embodiments the loudness modelprocessor 201 can select one from any number of potential model formssuch as the Zwicker loudness model discussed in Launer's dissertation(http://medi.uni-oldenburg.de/download/docs/diss/Launer_1995_LoudnessPerception/appendix.pdf).

In some embodiments the loudness model processor 201 can select a commonloudness model to apply to the input data according to the processingcapability or capacity of the apparatus 10.

The application of the loudness model and the estimation of the primaryloudness characteristic or value is shown in FIG. 9 by step 801.

In some embodiments the loudness estimator comprises a switch 202. Theprimary loudness estimate in such embodiments is output to the switch202 which can in some embodiments can output the primary loudnessestimate to at least one of the ‘mode’ or ‘environment modifier’pathways dependent on the environmental condition the apparatus isoperating in or the mode of the operation of the apparatus. For examplein some embodiments the switch 202 can output the primary loudnessestimate to at least one acoustic modification models to process andmodify the primary loudness value to determine a condition or modemodified loudness estimate for the audio frame. For example in someembodiments the modes of operation or ‘environmental conditions’ can be‘headphone connection’, ‘hands-free connection’, or ‘earpiececonnection’ and used to indicate how the apparatus is being used.

In other words the apparatus may have the capacity to receiveheadphones, have the capacity to generate the audio output using aninternal hands free loudspeaker, or be capable of outputting the audiosignal via an earpiece loudspeaker.

The selection of one of the acoustics models is shown in FIG. 9 by step803.

In such embodiments the switch 202 can output the loudness primaryestimate to the headphone pathway. The headphone pathway comprises amaximiser 207 which monitors and stores a temporary maximum value beforeoutputting the value on the headphones output (L_(headphones)).

In some embodiments, for example those where a loudness estimate ispre-determined or pre-calculated and used later the loudness estimatescan be determined and then the maximiser 207 output a maximum value forthe file or audio track which cant then be stored in memory. In theseembodiments when the same track/file is played the pre-calculated valuecan be retrieved from memory and input to the parameter updater, forexample the parameter updated as shown in FIG. 5 and described infurther detail later.

In some embodiments for example those embodiments where the loudnessestimate is determined when the file/track is to be played or whilebeing placed, the pathways do not comprise maximisers to monitor amaximum value for the track/file and can in some embodiments be passedto the smoother as shown in FIG. 2 and described also in further detaillater.

In some embodiments the apparatus can on detecting a headphone setinteracting with the apparatus cause the switch to select the headphonepathway to receive the loudness primary estimate.

In some embodiments the apparatus can further determine the type ormodel of the headphone interacting with the apparatus and can in theseembodiments further modify the loudness estimate before the maximiser207 to simulate the acoustic effect of the type or model of theapparatus being operated.

Furthermore in some embodiments the switch 202 can select the secondpathway, the integrated hands free loudspeaker acoustics model pathwayon determining that the apparatus is operating in a hands free modeusing the integrated hands free loudspeaker. In such embodiments theloudspeaker and acoustics model pathway comprises an IHF loudspeaker andacoustics model 203 which can be a filter which simulates the acousticeffect of the IHF loudspeaker characteristics and the acousticalproperties of the IHF surroundings. The output of the IHF loudspeakerand acoustics model 203 filter can then be output to a IHF pathwaymaximiser 209 which stores a temporary maximum value of the loudnessvalue and outputs this on a loudness IHF (L_(IHF)) output.

Furthermore in some embodiments the switch outputs the primary loudnessvalue received from the loudness model processor 201 to an earpieceloudspeaker and acoustics model pathway on determining that theapparatus is operating using the integrated earpiece. The earpieceloudspeaker and acoustics model pathway in such embodiments can comprisean earpiece loudspeaker and acoustics model processor 205 which can be afilter simulating the acoustical effect of the earpiece loudspeaker. Theoutput of the earpiece loudspeaker and acoustics model 205 filter can beoutput to a third maximiser 211 which stores a temporary maximum valuebefore outputting on a earpiece loudness (L_(earpiece)) output.

The application of an acoustics model to generate a modified loudnessestimate is shown in FIG. 9 by step 805. However it would be appreciatedthat in some embodiments the acoustics model may be a ‘null’ effect. Inother words the loudness estimate modification acoustic model may outputthe same loudness estimate value as was input.

In some embodiments as described above the switch 202 is configured tooutput the loudness model output to more than one pathway and then atleast one output of all the pathways as selected to be output.

In some embodiments the loudness balancer 199 further comprises asmoother 105. The smoother in such embodiments attempts to preventsudden changes in the gain changes by smoothing the loudness estimateswhich can vary significantly within a single track and cause overreactive changes in the later parameter determination. In someembodiments the smoother 105 can comprise a low pass filter whichreceives at least one of the loudness estimates and smoothes the segmentby segment loudness estimation values. In such embodiments the loudnessestimates missing for example when a silence or non-speech frame isdetermined in downlink communication examples can be allowed for. Thesmoothing of the modified loudness estimates as shown in FIG. 9 as step807.

Furthermore the smoothing of the loudness estimation is shown in FIG. 7by step 607.

In some embodiments the balancer further comprises a parameter updater107. The parameter updater 107 receives the smoothed loudness estimationvalues and furthermore receives a target loudness level input which maybe selected by the user, for example may be selected by the user by theuse of a ‘volume’ input or setting. In some other embodiments the targetloudness level input can be set either automatically orsemi-automatically. In some embodiments the target loudness level inputcan be generated by the user interface of the apparatus (enteredmanually or semi-automatically from a range of values). The parameterupdater 107 furthermore is configured to generate a parameter for thesignal pathway which may be used to control the processing of the audiosignal. For example the processing parameter may be a gain or dynamiclevel controller parameter. The controlling/determination of at leastone parameter for loudness control is shown in FIG. 7 by step 609.

Furthermore with respect to FIG. 5 a parameter updater 107 according tosome embodiments of the application is shown in further detail.Furthermore with respect to FIG. 10 the operation of such a parameterupdater 107 is shown in further detail. In some embodiments theparameter updater 107 comprises a loudness comparator 401 configured tocompare the loudness estimates and target loudness level input and aparameter controller 403 configured to generate control signalsdependent on the output from the loudness comparator 401.

The loudness estimates from the smoother 105, in other words thesmoothed L_(headphones)/L_(IHF)/L_(earpiece) values, are passed in someembodiments to the loudness comparator 401. Furthermore the loudnesscomparator 401 in such embodiments receives the target loudness levelinput. The loudness comparator 401 in such embodiments compares thesmoothed estimated loudness value and the target loudness level input todetermine an indicator or error value which can be output to theparameter controller 403. In some embodiments the loudness comparator401 generates an expected difference between the target loudness levelinput and the smoothed estimated loudness estimate which is passed tothe parameter controller 403.

The comparison between the smoothed estimated loudness value with thetarget loudness level input is shown in FIG. 10 by step 901.

Furthermore as described above in some embodiments the parameter updater107 further comprises the parameter controller 403 configured to recedethe output of the loudness comparator 401 and further configured togenerate a parameter value to control at least part of the signalpathway. In the above examples the control of the signal pathway isshown by the passing of a control parameter to an amplifier 109, howeveras previously discussed the signal pathway processing element can be anysuitable processing of the audio signal of which gain values may bemanipulated. For example in some embodiments a dynamic level controller(DLC) or dynamic range controller (DRC) can be implemented to controlthe audio signal.

The parameter controller 403 in some embodiments can generate thecontrol parameters in some embodiments by using a look-up table whichoutputs a gain parameter dependent on the error value such that the gainis increased when the target loudness level input greater than thesmoothed estimated loudness value but decreases the gain value when thetarget loudness level input is less than the smoothed estimated loudnessvalue.

Generation of at least one control parameter for the signal pathwayprocessing is shown in FIG. 10 by step 903.

Furthermore with respect to the summary operation FIG. 7 the applicationof at least one control parameter for loudness control is shown in FIG.7 by step 609.

In some embodiments, as discussed above, the loudness balancer 199 canbe considered to compose an amplifier 109 in the signal pathway. Theamplifier 109 in such embodiments receives the audio signal and the gainparameter from the parameter updater 107 and amplifies the audio signaldependent on the gain parameter such that when the smoothed estimateloudness value is below the target loudness level input, theamplification is increased to balance the level of the loudness.Similarly in such embodiments the amplifier 109 is configured todecrease the amplification when the smoothed estimate loudness value isabove the target loudness level input.

The application of the at least one control parameter to theamplifier/dynamic level controller is shown in FIG. 10 by step 905.

Furthermore the operation of modifying the input audio signal using theat least one control parameter is shown in FIG. 7 by step 611.

Therefore in embodiments such as described above, once the loudnessestimation has been calculated the audio signal processing (such as adynamic range controller (DRC) or dynamic level controller (DLC) or anamplifier may be adaptively adjusted to produce a consistent perceivedloudness for the user relevant to the target loudness requested by theuser. Thus where the loudness model identifies a severe reduction orincrease of loudness values as compared to a default setting for anaudio track the loudness balancer 199 can update parameters, which insome embodiments may be software parameters, such as dynamic rangecontroller values or gain values. For example the dynamic rangecontroller can in some embodiments be tuned in light of a particulartest signal where the characteristics of a test signal could be comparedagainst a source signal such as a music track.

In some embodiments this comparison could be a threshold detection.Furthermore in some embodiments the system could calculate loudnessestimates for all types of signals (music tracks) as soon as they areplaced in the music library in advance and therefore the parameterscould be updated as soon as the user selects playing the music trackfrom the library.

In other embodiments the playback system in the apparatus uses a shortsilent portion, for example the 2 or 3 seconds between each music track,and during the silent portion the system calculates the loudnessestimate and decides whether the DSP parameters or amplifier valuesrequire updating and if so updates the DSP parameters or amplificationvalue in order that the loudness level produced is consistent with theexpected or required loudness value.

In some other embodiments in order to reduce the effect of theadditional break between tracks where predefined values are notavailable in some embodiments the analysis can be carried out in realtime with a very slow smoothing filter applied. Furthermore in suchembodiments as the track or audio file is being played for the firsttime the determined loudness estimate values may be determined andstored for future plays of the same track or audio file.

In some embodiments determination of the new gain value or DSPparameters generated by the parameter updater uses an adaptive algorithmsuch that the error term produced by the comparator between the targetand estimated loudness levels changes with step size depending on thepercentage or value of the error value compared to the target value.

Furthermore in some embodiments where the signal path uses a dynamicrange controller the parameter updater 107 can control the dynamic rangecontrol parameters such as the compression ratio. In such embodimentsthe signal path can further comprise a loudness normalisation gainoperation. In some embodiments the audio loudness balancing can befurther improved by introducing pre-processing of the audio tracks orfiles. For example in some embodiments audio visual files, such as musicand videos which are stored in the apparatus' internal memory comprise apre-processor configured to determine a single loudness value for thespecific file or track and can store this single value in memoryassociated with the audio track or file. In such embodiments the singleloudness value can be the peak loudness value within the file which isstored onto a database and the specific track or file. Furthermore insuch embodiments the parameter updater 107 can receive the pre-processedloudness value corresponding to the track to be played. Thispre-processed value in some embodiments may then be modified accordingto the mode of operation/environmental conditions of the apparatus. Insome embodiments more than one loudness value can be determined storedand used, for example the pre-processor in some embodiments can beconfigured to determine loudness values for the file/track associatedwith IHF, earpiece and headphone loudness values. These values can thenin these embodiments be stored for each track in order that the levelmay be adjusted according to the environment within which the track isbeing played.

In some embodiments a further feedback loop may be implemented in orderto improve the loudness balancing. With respect to FIG. 6 and FIG. 11,apparatus according to these embodiments and additional operations ofthese embodiments compared with the embodiments described previously isshown in further detail.

The example apparatus shown in FIG. 6 is similar to the exampleapparatus shown in FIG. 2 but further comprises a feedback loop used toassist controlling the loudness balancing operation. It would beappreciated that the feedback loop described hereafter may beimplemented in combination with any of the previously describedembodiments and not only with the example apparatus shown in FIG. 2 anddescribed previously.

The loudness balancer in FIG. 6 further comprises a microphone 303 whichis configured to receive audio input from the environment. In someembodiments the microphone 303 is configured to output a digital formatsignal, and thus may be an integrated filter comprising an analogue todigital converter of a suitable type. In some other embodiments themicrophone is configured to output an analogue signal to a suitableanalogue-to-digital converter (ADC) the output of which is then furtherprocessed according to the following description.

The operation of receiving the audio input by the microphone is shown inFIG. 11 by step 1001.

Furthermore the loudness balancer further comprises in these embodimentsa filter 305 configured to filter the microphone signal using a transferfunction representing the inverse of the path from thedigital-to-analogue converter (DAC) to the analogue-to-digital converter(ADC) path function. In other words the filter is configured to attemptto remove any biasing caused by the digital-to-analogue conversionfollowing the amplifier 109, the microphone analogue-to-digitalconversion described above, and also allow for any acoustic effect ofthe loudspeaker and the mechanical arrangement of the loudspeaker. Theoutput of the filter 305 is passed to a microphone root mean square(RMS) processor 307.

The filtering of the microphone signal is shown in FIG. 11 by step 1003.

The loudness balancer further in these embodiments comprises amicrophone RMS processor 307 configured to receive the output of thefilter 305, perform a root mean square operation on the filteredmicrophone signal, and output the root mean squared microphone value toa ratio processor 309. The root mean square (RMS) value may becalculated using any suitable root mean square process.

The determination of the root mean square of the microphone output isshown in FIG. 11 by step 1005.

The loudness balancer is these embodiments further comprises a audiosignal RMS processor 301 configured to receive the output of the signalpath, which is in this example represented by the amplifier 109, performa root mean square operation on the audio signal output by the signalpath, and output the root mean squared audio signal value to the ratioprocessor 309.

The determination of the root mean square of the signal pathway(amplifier/dynamic level dynamic range controller) output is shown inFIG. 11 by step 1006.

The loudness balancer is these embodiments further comprises a ratiocombiner which is configured to output a ratio of the two received RMSvalues to a multiplier 311.

The operation of determining the ratio of the microphone RMS to thesignal pathway RMS value is shown in FIG. 11 by step 1007.

The ratio thus can be considered to be a determination of the actualloudness or the loudness experienced by the microphone against theoutput loudness values passed to the transducer.

In some embodiments the loudness balancer further comprises a multiplier311.

The multiplier 311 in these embodiments is configured to receive theoutput of the loudness estimate and the ratio combiner, generate aproduct of the estimated loudness and ratio and pass the multipliedloudness estimate to the smoother 105 which smoothes the value asdescribed above. In other words the feedback loop attempts to compensatefor the loudness animation model against any bias in the loudnessestimation. The operation of generating the product of the ratio to theestimated loudness value is shown in FIG. 11 by step 1009.

Although we have described the operation of the analyser as beingcarried out with regards to the frequency domain, it would be understooda time domain implementation where sub-band time domain filtering andintegration may be carried out in order to produce similar outputresults.

In some embodiments the loudness balancer 199 can further comprise aplayback compensator. The playback compensator can in these embodimentsbe configured to allow for or compensate for the any processing effecton the processing path or signal pathway which affects the loudnessestimation other than the parameter control affects. Thus for examplewhere the processing effect was one of a spectral processing orequalization the parameters of the various spectral gains could bepassed to the loudness balancer playback compensator to determine theeffect of the processing on the loudness estimation for the track/filebeing analysed and played.

Thus in summary at least one embodiment of the application performs amethod comprising: determining a loudness estimate of a first audiosignal; generating a parameter dependent on the loudness estimate; andcontrolling the first audio signal dependent on the parameter.

Although the above examples describe embodiments of the inventionoperating within an electronic device 10 or apparatus, it would beappreciated that the invention as described below may be implemented aspart of any audio processor. Thus, for example, embodiments of theinvention may be implemented in an audio processor which may implementaudio processing over fixed or wired communication paths.

Thus user equipment may comprise an audio processor such as thosedescribed in embodiments of the invention above.

It shall be appreciated that the term electronic device and userequipment is intended to cover any suitable type of wireless userequipment, such as mobile telephones, portable data processing devicesor portable web browsers.

In general, the various embodiments of the invention may be implementedin hardware or special purpose circuits, software, logic or anycombination thereof. For example, some aspects may be implemented inhardware, while other aspects may be implemented in firmware or softwarewhich may be executed by a controller, microprocessor or other computingdevice, although the invention is not limited thereto. While variousaspects of the invention may be illustrated and described as blockdiagrams, flow charts, or using some other pictorial representation, itis well understood that these blocks, apparatus, systems, techniques ormethods described herein may be implemented in, as non-limitingexamples, hardware, software, firmware, special purpose circuits orlogic, general purpose hardware or controller or other computingdevices, or some combination thereof.

Thus at least some embodiments may be an apparatus comprising at leastone processor and at least one memory including computer program codethe at least one memory and the computer program code configured to,with the at least one processor, cause the apparatus at least toperform: determining a loudness estimate of a first audio signal;generating a parameter dependent on the loudness estimate; andcontrolling the first audio signal dependent on the parameter.

The embodiments of this invention may be implemented by computersoftware executable by a data processor of the mobile device, such as inthe processor entity, or by hardware, or by a combination of softwareand hardware. Further in this ragard it should be noted that any blocksof the logic flow as in the Figures may represent program steps, orinterconnected logic ciruits, blocks and functions, or a combination ofprogram steps and logic circuits, blocks and functions. The software maybe stored on such physical media as memory chips, or memory blocksimplemented within the processor, magnetic media such as hard disk orfloppy disks, and optical media such as for example DVD and the datavariants thereof, CD.

Thus at least some embodiments may be a computer-readable medium encodedwith instructions that, when executed by a computer perform: determininga loudness estimate of a first audio signal; generating a parameterdependent on the loudness estimate; and controlling the first audiosignal dependent on the parameter.

The memory may be of any type suitable to the local technicalenvironment and may be implemented using any suitable data storagetechnology, such as semiconductor-based memory devices, magnetic memorydevices and systems, optical memory devices and systems, fixed memoryand removable memory. The data processors may be of any type suitable tothe local technical environment, and may include one or more of generalpurpose computers, special purpose computers, microprocessors, digitalsignal processors (DSPs), application specific integrated circuits(ASIC), gate level circuits and processors based on multi-core processorarchitecture, as non-limiting examples.

Embodiments of the invention may be practiced in various components suchas integrated circuit modules. The design of integrated circuits is byand large a highly automated process. Complex and powerful softwaretools are available for converting a logic level design into asemiconductor circuit design ready to be etched and formed on asemiconductor substrate.

Programs, such as those provided by Synopsys, Inc. of Mountain View,Calif. and Cadence Design, of San Jose, Calif. automatically routeconductors and locate components on a semiconductor chip using wellestablished rules of design as well as libraries of pre-stored designmodules. Once the design for a semiconductor circuit has been completed,the resultant design, in a standardized electronic format (e.g., Opus,GDSII, or the like) may be transmitted to a semiconductor fabricationfacility or “fab” for fabrication.

As used in this application, the term ‘circuitry’ refers to all of thefollowing:

-   -   (a) hardware-only circuit implementations (such as        implementations in only analog and/or digital circuitry) and    -   (b) to combinations of circuits and software (and/or firmware),        such as: (i) to a combination of processor(s)), software, and        memory(ies) that work together to cause an apparatus, such as a        mobile phone or server, to perform various functions and    -   (c) to circuits, such as a microprocessor(s) or a portion of a        microprocessor(s), that require software or firmware for        operation, even if the software or firmware is not physically        present.

This definition of ‘circuitry’ applies to all uses of this term in thisapplication, including any claims. As a further example, as used in thisapplication, the term ‘circuitry’ would also cover an implementation ofmerely a processor (or multiple processors) or portion of a processorand its (or their) accompanying software and/or firmware. The term‘circuitry’ would also cover, for example and if applicable to theparticular claim element, a baseband integrated circuit or applicationsprocessor integrated circuit for a mobile phone or similar integratedcircuit in server, a cellular network device, or other network device.

The foregoing description has provided by way of exemplary andnon-limiting examples a full and informative description of theexemplary embodiment of this invention. However, various modificationsand adaptations may become apparent to those skilled in the relevantarts in view of the foregoing description, when read in conjunction withthe accompanying drawings and the appended claims. However, all such andsimilar modifications of the teachings of this invention will still fallwithin the scope of this invention as defined in the appended claims.

The invention claimed is:
 1. A method comprising: receiving anindication for a mode of operation of an apparatus, wherein the mode ofoperation is determined at least in part by whether the apparatus isoperating in at least one of a headphone connection, a hands-freeconnection, or an earpiece connection; determining an audio signal isreceived during the mode of operation, the audio signal comprising atleast one of a speech signal or a non-speech signal, wherein thenon-speech signal comprises a music audio signal; determining arespective loudness level for playback for each of the speech signal andnon-speech signal of the audio signal based on whether the apparatus isoperating in the headphone connection, the hands-free connection, or theearpiece connection during the mode of operation; determining arespective loudness estimate based on the signal type of the audiosignal being at least one of the speech signal or the music audiosignal, wherein determining the respective loudness estimate comprisesat least applying at least one respective loudness model to the signaltype of the audio signal; generating at least one respective signalprocessing parameter dependent on the respective loudness estimate; andadjusting the respective signal type of the audio signal with therespective at least one signal processing parameter so as to update thedetermined respective loudness level.
 2. The method as claimed in claim1, wherein the at least one respective loudness model comprises at leastone of: a digital filter; or a parametric filter.
 3. The method asclaimed in claim 1, wherein the at least one respective loudness modelcomprises a speech loudness model to be applied to the audio signal whenthe audio signal comprises a speech frame.
 4. The method as claimed inclaim 1, wherein determining the respective loudness estimate furthercomprises applying an environmental model.
 5. The method as claimed inclaim 4, wherein the environmental model comprises at least one of: aloudspeaker model comprising at least one filter simulating an acousticeffect of an integrated hands free loudspeaker; an earpiece modelcomprising at least one filter simulating an acoustic effect of anearpiece transducer and housing; a headset model comprising at least onefilter simulating an acoustic effect of a predetermined headset; or amechanical model comprising at least one filter simulating a mechanicalacoustical response of the apparatus.
 6. The method as claimed in claim1, wherein the at least one respective signal processing parametercomprises at least one of: a gain control parameter applied to anamplifier configured to control the audio signal; a dynamic rangeparameter applied to a dynamic range processor configured to control theaudio signal; or a dynamic level parameter applied to a dynamic levelcontroller configured to control the audio signal.
 7. The method asclaimed in claim 1, wherein generating at least one respective signalprocessing parameter dependent on the respective loudness estimatecomprises: comparing the respective loudness estimate with a loudnessinput value; and generating the at least one respective signalprocessing parameter dependent on a difference between the respectiveloudness estimate of the audio signal and the loudness input value. 8.An apparatus comprising at least one processor and at least one memoryincluding computer code for one or more programs, the at least onememory and the computer code configured to with the at least oneprocessor cause the apparatus to: receive an indication for a mode ofoperation of the apparatus, wherein the mode of operation is determinedat least in part by whether the apparatus is operating in at least oneof a headphone connection, a hands-free connection, or an earpiececonnection; determine an audio signal is received during the mode ofoperation, the audio signal comprising at least one of a speech signalor a non-speech signal, wherein the non-speech signal comprises a musicaudio signal; determine a respective loudness level for playback foreach of the speech signal and non-speech signal of the audio signalbased on whether the apparatus is operating in the headphone connection,the hands-free connection, or the earpiece connection during the mode ofoperation; determine a respective loudness estimate based on the signaltype of the audio signal being at least one of the speech signal or themusic audio signal, wherein determining the respective loudness estimatecomprises at least applying at least one respective loudness model tothe signal type of the audio signal; generate at least one respectivesignal processing parameter dependent on the respective loudnessestimate; and adjust the respective signal type of the audio signal withthe respective at least one signal processing parameter so as to updatethe determined respective loudness level.
 9. The apparatus as claimed inclaim 8, wherein the at least one respective loudness model comprises atleast one of: a digital filter; or a parametric filter.
 10. Theapparatus as claimed in claim 8, wherein the at least one respectiveloudness model comprises a speech loudness model to be applied to theaudio signal when the audio signal comprises a speech frame.
 11. Theapparatus as claimed in claim 8, wherein determining the respectiveloudness estimate further comprises applying an environmental model. 12.The apparatus as claimed in claim 11, wherein the environmental modelcomprises at least one of: a loudspeaker model comprising at least onefilter simulating an acoustic effect of an integrated hands freeloudspeaker; an earpiece model comprising at least one filter simulatingan acoustic effect of an earpiece transducer and housing; a headsetmodel comprising at least one filter simulating an acoustic effect of apredetermined headset; or a mechanical model comprising at least onefilter simulating a mechanical acoustical response of the apparatus. 13.The apparatus as claimed in claim 8, wherein the at least one respectivesignal processing parameter comprises at least one of: a gain controlparameter applied to an amplifier configured to control the audiosignal; a dynamic range parameter applied to a dynamic range processorconfigured to control the audio signal; or a dynamic level parameterapplied to a dynamic level controller configured to control the audiosignal.
 14. The apparatus as claimed in claim 8, wherein generating atleast one respective signal processing parameter dependent on therespective loudness estimate comprises: comparing the respectiveloudness estimate with a loudness input value; and generating the atleast one respective signal processing parameter dependent on adifference between the respective loudness estimate of the audio signaland the loudness input value.
 15. A computer program product comprisingat least one non-transitory computer-readable storage medium havingcomputer-executable program code portions stored therein, thecomputer-executable program code portions comprising program codeinstructions for: receiving an indication for a mode of operation of anapparatus, wherein the mode of operation is determined at least in partby whether the apparatus is operating in at least one of a headphoneconnection, a hands-free connection, or an earpiece connection;determining an audio signal is received during the mode of operation,the audio signal comprising at least one of a speech signal or anon-speech signal, wherein the non-speech signal comprises a music audiosignal; determining a respective loudness level for playback for each ofthe speech signal and non-speech signal of the audio signal based onwhether the apparatus is operating in the headphone connection, thehands-free connection, or the earpiece connection during the mode ofoperation; determining a respective loudness estimate based on thesignal type of the audio signal being at least one of the speech signalor the music audio signal, wherein determining the respective loudnessestimate comprises at least applying at least one respective loudnessmodel to the signal type of the audio signal; generating at least onerespective signal processing parameter dependent on the respectiveloudness estimate; and adjusting the respective signal type of the audiosignal with the respective at least one signal processing parameter soas to update the determined respective loudness level.
 16. The computerprogram product as claimed in claim 15, wherein the at least onerespective loudness model comprises at least one of: a digital filter;or a parametric filter.
 17. The computer program product as claimed inclaim 15, wherein the at least one respective loudness model comprises aspeech loudness model to be applied to the audio signal when the audiosignal comprises a speech frame.
 18. The computer program product asclaimed in claim 15, wherein determining the respective loudnessestimate further comprises applying an environmental model.
 19. Thecomputer program product as claimed in claim 18, wherein theenvironmental model comprises at least one of: a loudspeaker modelcomprising at least one filter simulating an acoustic effect of anintegrated hands free loudspeaker; an earpiece model comprising at leastone filter simulating an acoustic effect of an earpiece transducer andhousing; a headset model comprising at least one filter simulating anacoustic effect of a predetermined headset; or a mechanical modelcomprising at least one filter simulating a mechanical acousticalresponse of the apparatus.
 20. The computer program product as claimedin claim 15, wherein the at least one respective signal processingparameter comprises at least one of: a gain control parameter applied toan amplifier configured to control the audio signal; a dynamic rangeparameter applied to a dynamic range processor configured to control theaudio signal; or a dynamic level parameter applied to a dynamic levelcontroller configured to control the audio signal.